aaudio: convert mono output to stereo
MMAP hardware streams are often stereo.
But apps often want to output a mono stream.
Since converting from mono to stereo is easy,
we can go ahead and open a stereo hardware stream
and then just convert the apps mono data to stereo for the HW.
Add getDeviceChannelCount().
Test: adb shell write_sine_callback -pl -s10 -c1 -m3
Test: adb shell write_sine_callback -pl -s10 -c1 -m3 -x
Change-Id: I444a38c6f5cd32d1d6113f16aacec68285a1bc82
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index b611160..6b25302 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -104,7 +104,7 @@
request.setUserId(getuid());
request.setProcessId(getpid());
request.setSharingModeMatchRequired(isSharingModeMatchRequired());
- request.setInService(mInService);
+ request.setInService(isInService());
request.getConfiguration().setDeviceId(getDeviceId());
request.getConfiguration().setSampleRate(getSampleRate());
@@ -118,11 +118,24 @@
request.getConfiguration().setBufferCapacity(builder.getBufferCapacity());
+ mDeviceChannelCount = getSamplesPerFrame(); // Assume it will be the same. Update if not.
+
mServiceStreamHandle = mServiceInterface.openStream(request, configurationOutput);
+ if (mServiceStreamHandle < 0
+ && request.getConfiguration().getSamplesPerFrame() == 1 // mono?
+ && getDirection() == AAUDIO_DIRECTION_OUTPUT
+ && !isInService()) {
+ // if that failed then try switching from mono to stereo if OUTPUT.
+ // Only do this in the client. Otherwise we end up with a mono mixer in the service
+ // that writes to a stereo MMAP stream.
+ ALOGD("%s - openStream() returned %d, try switching from MONO to STEREO",
+ __func__, mServiceStreamHandle);
+ request.getConfiguration().setSamplesPerFrame(2); // stereo
+ mServiceStreamHandle = mServiceInterface.openStream(request, configurationOutput);
+ }
if (mServiceStreamHandle < 0) {
- result = mServiceStreamHandle;
- ALOGE("%s - openStream() returned %d", __func__, result);
- return result;
+ ALOGE("%s - openStream() returned %d", __func__, mServiceStreamHandle);
+ return mServiceStreamHandle;
}
result = configurationOutput.validate();
@@ -130,8 +143,12 @@
goto error;
}
// Save results of the open.
+ if (getSamplesPerFrame() == AAUDIO_UNSPECIFIED) {
+ setSamplesPerFrame(configurationOutput.getSamplesPerFrame());
+ }
+ mDeviceChannelCount = configurationOutput.getSamplesPerFrame();
+
setSampleRate(configurationOutput.getSampleRate());
- setSamplesPerFrame(configurationOutput.getSamplesPerFrame());
setDeviceId(configurationOutput.getDeviceId());
setSessionId(configurationOutput.getSessionId());
setSharingMode(configurationOutput.getSharingMode());
@@ -160,7 +177,6 @@
goto error;
}
-
// Validate result from server.
framesPerBurst = mEndpointDescriptor.dataQueueDescriptor.framesPerBurst;
if (framesPerBurst < MIN_FRAMES_PER_BURST || framesPerBurst > MAX_FRAMES_PER_BURST) {
diff --git a/media/libaaudio/src/client/AudioStreamInternal.h b/media/libaaudio/src/client/AudioStreamInternal.h
index 0f54f8c..0e0724b 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.h
+++ b/media/libaaudio/src/client/AudioStreamInternal.h
@@ -138,7 +138,14 @@
// Calculate timeout for an operation involving framesPerOperation.
int64_t calculateReasonableTimeout(int32_t framesPerOperation);
- aaudio_format_t mDeviceFormat = AAUDIO_FORMAT_UNSPECIFIED;
+ aaudio_format_t getDeviceFormat() const { return mDeviceFormat; }
+
+ int32_t getDeviceChannelCount() const { return mDeviceChannelCount; }
+
+ /**
+ * @return true if running in audio service, versus in app process
+ */
+ bool isInService() const { return mInService; }
IsochronousClockModel mClockModel; // timing model for chasing the HAL
@@ -187,6 +194,11 @@
EndpointDescriptor mEndpointDescriptor; // buffer description with resolved addresses
int64_t mServiceLatencyNanos = 0;
+
+ // Sometimes the hardware is operating with a different format or channel count from the app.
+ // Then we require conversion in AAudio.
+ aaudio_format_t mDeviceFormat = AAUDIO_FORMAT_UNSPECIFIED;
+ int32_t mDeviceChannelCount = 0;
};
} /* namespace aaudio */
diff --git a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
index 62f0fc8..0719fe1 100644
--- a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
@@ -176,16 +176,16 @@
int32_t numSamples = framesToProcess * getSamplesPerFrame();
// TODO factor this out into a utility function
- if (mDeviceFormat == getFormat()) {
+ if (getDeviceFormat() == getFormat()) {
memcpy(destination, wrappingBuffer.data[partIndex], numBytes);
- } else if (mDeviceFormat == AAUDIO_FORMAT_PCM_I16
+ } else if (getDeviceFormat() == AAUDIO_FORMAT_PCM_I16
&& getFormat() == AAUDIO_FORMAT_PCM_FLOAT) {
AAudioConvert_pcm16ToFloat(
(const int16_t *) wrappingBuffer.data[partIndex],
(float *) destination,
numSamples,
1.0f);
- } else if (mDeviceFormat == AAUDIO_FORMAT_PCM_FLOAT
+ } else if (getDeviceFormat() == AAUDIO_FORMAT_PCM_FLOAT
&& getFormat() == AAUDIO_FORMAT_PCM_I16) {
AAudioConvert_floatToPcm16(
(const float *) wrappingBuffer.data[partIndex],
diff --git a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
index 5660c1b..7695dfa 100644
--- a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
@@ -194,7 +194,7 @@
// ALOGD("AudioStreamInternal::writeNowWithConversion(%p, %d)",
// buffer, numFrames);
WrappingBuffer wrappingBuffer;
- uint8_t *source = (uint8_t *) buffer;
+ uint8_t *byteBuffer = (uint8_t *) buffer;
int32_t framesLeft = numFrames;
mAudioEndpoint.getEmptyFramesAvailable(&wrappingBuffer);
@@ -208,69 +208,26 @@
if (framesToWrite > framesAvailable) {
framesToWrite = framesAvailable;
}
+
int32_t numBytes = getBytesPerFrame() * framesToWrite;
- int32_t numSamples = framesToWrite * getSamplesPerFrame();
// Data conversion.
float levelFrom;
float levelTo;
- bool ramping = mVolumeRamp.nextSegment(framesToWrite, &levelFrom, &levelTo);
- // The formats are validated when the stream is opened so we do not have to
- // check for illegal combinations here.
- // TODO factor this out into a utility function
- if (getFormat() == AAUDIO_FORMAT_PCM_FLOAT) {
- if (mDeviceFormat == AAUDIO_FORMAT_PCM_FLOAT) {
- AAudio_linearRamp(
- (const float *) source,
- (float *) wrappingBuffer.data[partIndex],
- framesToWrite,
- getSamplesPerFrame(),
- levelFrom,
- levelTo);
- } else if (mDeviceFormat == AAUDIO_FORMAT_PCM_I16) {
- if (ramping) {
- AAudioConvert_floatToPcm16(
- (const float *) source,
- (int16_t *) wrappingBuffer.data[partIndex],
- framesToWrite,
- getSamplesPerFrame(),
- levelFrom,
- levelTo);
- } else {
- AAudioConvert_floatToPcm16(
- (const float *) source,
- (int16_t *) wrappingBuffer.data[partIndex],
- numSamples,
- levelTo);
- }
- }
- } else if (getFormat() == AAUDIO_FORMAT_PCM_I16) {
- if (mDeviceFormat == AAUDIO_FORMAT_PCM_FLOAT) {
- if (ramping) {
- AAudioConvert_pcm16ToFloat(
- (const int16_t *) source,
- (float *) wrappingBuffer.data[partIndex],
- framesToWrite,
- getSamplesPerFrame(),
- levelFrom,
- levelTo);
- } else {
- AAudioConvert_pcm16ToFloat(
- (const int16_t *) source,
- (float *) wrappingBuffer.data[partIndex],
- numSamples,
- levelTo);
- }
- } else if (mDeviceFormat == AAUDIO_FORMAT_PCM_I16) {
- AAudio_linearRamp(
- (const int16_t *) source,
- (int16_t *) wrappingBuffer.data[partIndex],
- framesToWrite,
- getSamplesPerFrame(),
- levelFrom,
- levelTo);
- }
- }
- source += numBytes;
+ mVolumeRamp.nextSegment(framesToWrite, &levelFrom, &levelTo);
+
+ AAudioDataConverter::FormattedData source(
+ (void *)byteBuffer,
+ getFormat(),
+ getSamplesPerFrame());
+ AAudioDataConverter::FormattedData destination(
+ wrappingBuffer.data[partIndex],
+ getDeviceFormat(),
+ getDeviceChannelCount());
+
+ AAudioDataConverter::convert(source, destination, framesToWrite,
+ levelFrom, levelTo);
+
+ byteBuffer += numBytes;
framesLeft -= framesToWrite;
} else {
break;
diff --git a/media/libaaudio/src/utility/AAudioUtilities.cpp b/media/libaaudio/src/utility/AAudioUtilities.cpp
index 854c691..40b31b9 100644
--- a/media/libaaudio/src/utility/AAudioUtilities.cpp
+++ b/media/libaaudio/src/utility/AAudioUtilities.cpp
@@ -27,6 +27,7 @@
#include <aaudio/AAudioTesting.h>
#include <math.h>
#include <system/audio-base.h>
+#include <assert.h>
#include "utility/AAudioUtilities.h"
@@ -72,7 +73,7 @@
int16_t *destination,
int32_t numSamples,
float amplitude) {
- float scaler = amplitude;
+ const float scaler = amplitude;
for (int i = 0; i < numSamples; i++) {
float sample = *source++;
*destination++ = clipAndClampFloatToPcm16(sample, scaler);
@@ -103,7 +104,7 @@
float *destination,
int32_t numSamples,
float amplitude) {
- float scaler = amplitude / SHORT_SCALE;
+ const float scaler = amplitude / SHORT_SCALE;
for (int i = 0; i < numSamples; i++) {
destination[i] = source[i] * scaler;
}
@@ -117,7 +118,7 @@
float amplitude1,
float amplitude2) {
float scaler = amplitude1 / SHORT_SCALE;
- float delta = (amplitude2 - amplitude1) / (SHORT_SCALE * (float) numFrames);
+ const float delta = (amplitude2 - amplitude1) / (SHORT_SCALE * (float) numFrames);
for (int frameIndex = 0; frameIndex < numFrames; frameIndex++) {
for (int sampleIndex = 0; sampleIndex < samplesPerFrame; sampleIndex++) {
*destination++ = *source++ * scaler;
@@ -134,7 +135,7 @@
float amplitude1,
float amplitude2) {
float scaler = amplitude1;
- float delta = (amplitude2 - amplitude1) / numFrames;
+ const float delta = (amplitude2 - amplitude1) / numFrames;
for (int frameIndex = 0; frameIndex < numFrames; frameIndex++) {
for (int sampleIndex = 0; sampleIndex < samplesPerFrame; sampleIndex++) {
float sample = *source++;
@@ -158,7 +159,7 @@
float amplitude2) {
// Because we are converting from int16 to 1nt16, we do not have to scale by 1/32768.
float scaler = amplitude1;
- float delta = (amplitude2 - amplitude1) / numFrames;
+ const float delta = (amplitude2 - amplitude1) / numFrames;
for (int frameIndex = 0; frameIndex < numFrames; frameIndex++) {
for (int sampleIndex = 0; sampleIndex < samplesPerFrame; sampleIndex++) {
// No need to clip because int16_t range is inherently limited.
@@ -169,6 +170,255 @@
}
}
+// *************************************************************************************
+// Convert Mono To Stereo at the same time as converting format.
+void AAudioConvert_formatMonoToStereo(const float *source,
+ int16_t *destination,
+ int32_t numFrames,
+ float amplitude) {
+ const float scaler = amplitude;
+ for (int i = 0; i < numFrames; i++) {
+ float sample = *source++;
+ int16_t sample16 = clipAndClampFloatToPcm16(sample, scaler);
+ *destination++ = sample16;
+ *destination++ = sample16;
+ }
+}
+
+void AAudioConvert_formatMonoToStereo(const float *source,
+ int16_t *destination,
+ int32_t numFrames,
+ float amplitude1,
+ float amplitude2) {
+ // divide by numFrames so that we almost reach amplitude2
+ const float delta = (amplitude2 - amplitude1) / numFrames;
+ for (int frameIndex = 0; frameIndex < numFrames; frameIndex++) {
+ const float scaler = amplitude1 + (frameIndex * delta);
+ const float sample = *source++;
+ int16_t sample16 = clipAndClampFloatToPcm16(sample, scaler);
+ *destination++ = sample16;
+ *destination++ = sample16;
+ }
+}
+
+void AAudioConvert_formatMonoToStereo(const int16_t *source,
+ float *destination,
+ int32_t numFrames,
+ float amplitude) {
+ const float scaler = amplitude / SHORT_SCALE;
+ for (int i = 0; i < numFrames; i++) {
+ float sample = source[i] * scaler;
+ *destination++ = sample;
+ *destination++ = sample;
+ }
+}
+
+// This code assumes amplitude1 and amplitude2 are between 0.0 and 1.0
+void AAudioConvert_formatMonoToStereo(const int16_t *source,
+ float *destination,
+ int32_t numFrames,
+ float amplitude1,
+ float amplitude2) {
+ const float scaler1 = amplitude1 / SHORT_SCALE;
+ const float delta = (amplitude2 - amplitude1) / (SHORT_SCALE * (float) numFrames);
+ for (int frameIndex = 0; frameIndex < numFrames; frameIndex++) {
+ float scaler = scaler1 + (frameIndex * delta);
+ float sample = source[frameIndex] * scaler;
+ *destination++ = sample;
+ *destination++ = sample;
+ }
+}
+
+// This code assumes amplitude1 and amplitude2 are between 0.0 and 1.0
+void AAudio_linearRampMonoToStereo(const float *source,
+ float *destination,
+ int32_t numFrames,
+ float amplitude1,
+ float amplitude2) {
+ const float delta = (amplitude2 - amplitude1) / numFrames;
+ for (int frameIndex = 0; frameIndex < numFrames; frameIndex++) {
+ float sample = *source++;
+
+ // Clip to valid range of a float sample to prevent excessive volume.
+ if (sample > MAX_HEADROOM) sample = MAX_HEADROOM;
+ else if (sample < MIN_HEADROOM) sample = MIN_HEADROOM;
+
+ const float scaler = amplitude1 + (frameIndex * delta);
+ float sampleScaled = sample * scaler;
+ *destination++ = sampleScaled;
+ *destination++ = sampleScaled;
+ }
+}
+
+// This code assumes amplitude1 and amplitude2 are between 0.0 and 1.0
+void AAudio_linearRampMonoToStereo(const int16_t *source,
+ int16_t *destination,
+ int32_t numFrames,
+ float amplitude1,
+ float amplitude2) {
+ // Because we are converting from int16 to 1nt16, we do not have to scale by 1/32768.
+ const float delta = (amplitude2 - amplitude1) / numFrames;
+ for (int frameIndex = 0; frameIndex < numFrames; frameIndex++) {
+ const float scaler = amplitude1 + (frameIndex * delta);
+ // No need to clip because int16_t range is inherently limited.
+ const float sample = *source++ * scaler;
+ int16_t sample16 = (int16_t) roundf(sample);
+ *destination++ = sample16;
+ *destination++ = sample16;
+ }
+}
+
+// *************************************************************************************
+void AAudioDataConverter::convert(
+ const FormattedData &source,
+ const FormattedData &destination,
+ int32_t numFrames,
+ float levelFrom,
+ float levelTo) {
+
+ if (source.channelCount == 1 && destination.channelCount == 2) {
+ convertMonoToStereo(source,
+ destination,
+ numFrames,
+ levelFrom,
+ levelTo);
+ } else {
+ // We only support mono to stereo conversion. Otherwise source and destination
+ // must match.
+ assert(source.channelCount == destination.channelCount);
+ convertChannelsMatch(source,
+ destination,
+ numFrames,
+ levelFrom,
+ levelTo);
+ }
+}
+
+void AAudioDataConverter::convertMonoToStereo(
+ const FormattedData &source,
+ const FormattedData &destination,
+ int32_t numFrames,
+ float levelFrom,
+ float levelTo) {
+
+ // The formats are validated when the stream is opened so we do not have to
+ // check for illegal combinations here.
+ if (source.format == AAUDIO_FORMAT_PCM_FLOAT) {
+ if (destination.format == AAUDIO_FORMAT_PCM_FLOAT) {
+ AAudio_linearRampMonoToStereo(
+ (const float *) source.data,
+ (float *) destination.data,
+ numFrames,
+ levelFrom,
+ levelTo);
+ } else if (destination.format == AAUDIO_FORMAT_PCM_I16) {
+ if (levelFrom != levelTo) {
+ AAudioConvert_formatMonoToStereo(
+ (const float *) source.data,
+ (int16_t *) destination.data,
+ numFrames,
+ levelFrom,
+ levelTo);
+ } else {
+ AAudioConvert_formatMonoToStereo(
+ (const float *) source.data,
+ (int16_t *) destination.data,
+ numFrames,
+ levelTo);
+ }
+ }
+ } else if (source.format == AAUDIO_FORMAT_PCM_I16) {
+ if (destination.format == AAUDIO_FORMAT_PCM_FLOAT) {
+ if (levelFrom != levelTo) {
+ AAudioConvert_formatMonoToStereo(
+ (const int16_t *) source.data,
+ (float *) destination.data,
+ numFrames,
+ levelFrom,
+ levelTo);
+ } else {
+ AAudioConvert_formatMonoToStereo(
+ (const int16_t *) source.data,
+ (float *) destination.data,
+ numFrames,
+ levelTo);
+ }
+ } else if (destination.format == AAUDIO_FORMAT_PCM_I16) {
+ AAudio_linearRampMonoToStereo(
+ (const int16_t *) source.data,
+ (int16_t *) destination.data,
+ numFrames,
+ levelFrom,
+ levelTo);
+ }
+ }
+}
+
+void AAudioDataConverter::convertChannelsMatch(
+ const FormattedData &source,
+ const FormattedData &destination,
+ int32_t numFrames,
+ float levelFrom,
+ float levelTo) {
+ const int32_t numSamples = numFrames * source.channelCount;
+
+ // The formats are validated when the stream is opened so we do not have to
+ // check for illegal combinations here.
+ if (source.format == AAUDIO_FORMAT_PCM_FLOAT) {
+ if (destination.format == AAUDIO_FORMAT_PCM_FLOAT) {
+ AAudio_linearRamp(
+ (const float *) source.data,
+ (float *) destination.data,
+ numFrames,
+ source.channelCount,
+ levelFrom,
+ levelTo);
+ } else if (destination.format == AAUDIO_FORMAT_PCM_I16) {
+ if (levelFrom != levelTo) {
+ AAudioConvert_floatToPcm16(
+ (const float *) source.data,
+ (int16_t *) destination.data,
+ numFrames,
+ source.channelCount,
+ levelFrom,
+ levelTo);
+ } else {
+ AAudioConvert_floatToPcm16(
+ (const float *) source.data,
+ (int16_t *) destination.data,
+ numSamples,
+ levelTo);
+ }
+ }
+ } else if (source.format == AAUDIO_FORMAT_PCM_I16) {
+ if (destination.format == AAUDIO_FORMAT_PCM_FLOAT) {
+ if (levelFrom != levelTo) {
+ AAudioConvert_pcm16ToFloat(
+ (const int16_t *) source.data,
+ (float *) destination.data,
+ numFrames,
+ source.channelCount,
+ levelFrom,
+ levelTo);
+ } else {
+ AAudioConvert_pcm16ToFloat(
+ (const int16_t *) source.data,
+ (float *) destination.data,
+ numSamples,
+ levelTo);
+ }
+ } else if (destination.format == AAUDIO_FORMAT_PCM_I16) {
+ AAudio_linearRamp(
+ (const int16_t *) source.data,
+ (int16_t *) destination.data,
+ numFrames,
+ source.channelCount,
+ levelFrom,
+ levelTo);
+ }
+ }
+}
+
status_t AAudioConvert_aaudioToAndroidStatus(aaudio_result_t result) {
// This covers the case for AAUDIO_OK and for positive results.
if (result >= 0) {
diff --git a/media/libaaudio/src/utility/AAudioUtilities.h b/media/libaaudio/src/utility/AAudioUtilities.h
index dc6a671..cea88fb 100644
--- a/media/libaaudio/src/utility/AAudioUtilities.h
+++ b/media/libaaudio/src/utility/AAudioUtilities.h
@@ -159,6 +159,41 @@
float amplitude1,
float amplitude2);
+class AAudioDataConverter {
+public:
+
+ struct FormattedData {
+
+ FormattedData(void *data, aaudio_format_t format, int32_t channelCount)
+ : data(data)
+ , format(format)
+ , channelCount(channelCount) {}
+
+ const void *data = nullptr;
+ const aaudio_format_t format = AAUDIO_FORMAT_UNSPECIFIED;
+ const int32_t channelCount = 1;
+ };
+
+ static void convert(const FormattedData &source,
+ const FormattedData &destination,
+ int32_t numFrames,
+ float levelFrom,
+ float levelTo);
+
+private:
+ static void convertMonoToStereo(const FormattedData &source,
+ const FormattedData &destination,
+ int32_t numFrames,
+ float levelFrom,
+ float levelTo);
+
+ static void convertChannelsMatch(const FormattedData &source,
+ const FormattedData &destination,
+ int32_t numFrames,
+ float levelFrom,
+ float levelTo);
+};
+
/**
* Calculate the number of bytes and prevent numeric overflow.
* @param numFrames frame count
diff --git a/services/oboeservice/AAudioService.cpp b/services/oboeservice/AAudioService.cpp
index ac3202b..c708fee 100644
--- a/services/oboeservice/AAudioService.cpp
+++ b/services/oboeservice/AAudioService.cpp
@@ -32,7 +32,6 @@
#include "AAudioService.h"
#include "AAudioServiceStreamMMAP.h"
#include "AAudioServiceStreamShared.h"
-#include "AAudioServiceStreamMMAP.h"
#include "binding/IAAudioService.h"
#include "ServiceUtilities.h"